Your browser-based softphone just got a significant upgrade. VitXi WebRTC v4.6.0-2 introduces automated call routing, rich text messaging for internal conversations, a new internal call history engine, and a handful of stability improvements that teams running VitalPBX have been asking for.
Here is a full breakdown of everything that changed and what it means for your daily workflow.
Automated Routing: No More “App vs SIP” Guesswork
One of the most friction-reducing changes in this release: the manual “App vs SIP” selection is gone.
VitXi now automatically detects the destination and routes accordingly — internal VitXi users go through the application, while extensions and external numbers route via SIP. You don’t have to think about it. You just dial.
This is particularly useful in environments where users are switching between internal and external calls frequently. Fewer clicks, fewer errors, faster calls.
Rich Text Messaging for Internal Conversations
Internal chats between VitXi users now support full rich text formatting: bold, italic, lists, and tables. If your team uses VitXi as a communication hub alongside voice, this makes a meaningful difference for structured updates, instructions, or anything more complex than a quick note.
Alongside rich text, the messaging UI has been redesigned with a cleaner, more responsive interface. Useful new additions include Shift + Enter for line breaks (no more accidental sends), new filters for “Most Recent” and “Unread Only,” and overall improved keyboard navigation.
Internal Call History Engine & Call Notes
VitXi now maintains its own call history database — separate from the PBX-level CDR — which enables faster UI updates and opens the door to more advanced capabilities going forward.
The first feature built on top of it is Call Notes: you can now add notes directly within a call history entry and have them persist. This is immediately useful for support teams, sales reps, or anyone who needs context on past interactions without jumping to a separate CRM.
It also sets the foundation for future AI capabilities tied to call history, so expect more here in upcoming releases.
Advanced ICE Server Settings: Force TURN / Relay Only
For IT administrators managing VitXi in complex network environments, v4.6.0-2 adds support for a “Relay Only” (Force TURN) transport policy.
In standard WebRTC, media traffic tries to go peer-to-peer (direct). In restrictive networks — symmetric NAT environments, strict corporate firewalls, or certain VPN configurations — this fails or degrades. Force TURN routes all media through your TURN server instead, ensuring reliable audio and video regardless of network topology.
If you have been seeing connectivity issues in specific network segments, this setting is worth testing.
Enhanced Caller Identification
VitXi v4.6.0-2 adds support for two SIP header standards — P-Asserted-Identity and Remote-Party-ID — which improve caller name detection, including during early call states before the call fully connects.
Combined with Smart Contact Resolution, which automatically matches incoming and outgoing numbers against your personal contacts and PBX phonebooks to display names and avatars, you now get a much more complete picture of who is calling — even for numbers that haven’t been dialed before.
And if a number isn’t in your contacts yet, Quick-Add Contacts lets you save it instantly from the chat interface.
Queue Controls for Agents
Agents working call queues now have clearer visibility into their queue status and improved action buttons for pausing, resuming, and logging out. Small change, but it reduces the back-and-forth for agents managing their availability throughout a shift.
Video, Screen Sharing & Network Quality
Video calls and screen sharing received stability and rendering improvements in this release, with better use of available screen space. A new Network Quality Indicator provides real-time feedback during active calls so users can see connection health at a glance — useful for diagnosing audio issues before they escalate.
E.164 Number Standardization
International numbers are now enforced in E.164 format across VitXi. This prevents duplicate conversations caused by number format variations (e.g., +1-555-555-0100 vs. 15555550100) and ensures consistent routing behavior for numbers that span different formats or regions.
Bug Fixes
Several issues that have been affecting reliability are resolved in this release:
- Audio loss on multiple calls — Fixed cases where audio dropped on recovered or concurrent calls
- Call interruption on incoming call — Resolved active calls dropping when a second call was received
- Video toggle stability — Fixed camera reactivation issues and duplicate video stream rendering
- Conference participant management — Improved reliability when adding participants to an active audio conference
- Voicemail & call history sync — Enhanced visual voicemail reliability and fixed history cleanup sync issues
Update Today
VitXi WebRTC v4.6.0-2 is available now. If you’re running VitalPBX with Vitxi, update your instance and share your feedback in the VitalPBX community forum.
Not using VitXi yet? It’s VitalPBX’s browser-based WebRTC softphone — no app install required, works directly from any modern browser. Explore it as part of a 30-day free trial or schedule a demo to see it in action.
Frequently Asked Questions
What is VitXi WebRTC?
VitXi is VitalPBX’s browser-based WebRTC softphone. It lets users make and receive calls, send internal messages, share screens, and join video calls directly from any modern web browser — no desktop app or plugin required. It is included with VitalPBX and is accessible through the VitalPBX web interface.
What does “Relay Only” (Force TURN) mean in v4.6.0-2?
Force TURN is a transport policy that routes all WebRTC media traffic — audio and video — through your configured TURN server instead of attempting a direct peer-to-peer connection. This is useful in restrictive network environments such as symmetric NAT configurations, strict corporate firewalls, or VPN setups where direct WebRTC connections fail or produce poor quality. Enabling it trades slightly higher latency for dramatically improved reliability in those environments.
What are Call Notes and where do they appear?
Call Notes are annotations you can attach to any call record in VitXi’s internal call history. You add them directly within the call history entry, and they persist across sessions. They are useful for logging what was discussed, what follow-up is needed, or any context that makes the next interaction with that contact more productive.
Does the automated routing change in v4.6.0-2 affect how calls are billed or logged?
No. The change simply removes the manual “App vs SIP” selection from the user interface. VitXi now detects the destination automatically — internal VitXi users go through the app, external numbers and extensions route via SIP — exactly as a user would have selected manually. Billing, CDR logging, and trunk routing behavior are unchanged.
Is VitXi available on mobile devices?
VitXi is a WebRTC browser-based softphone and works on any modern desktop or mobile browser. VitalPBX also offers VitalPBX Connect, a dedicated mobile app available on iOS and Android, included in the VitalPBX One plan. For heavy mobile use, VitalPBX Connect typically delivers a better native experience.
What VitalPBX plan includes VitXi?
VitXi is part of VitalPBX and is available across all commercial plans — VitalPBX One, Enterprise, Call Center, and Multi-Tenant. The free Community plan (up to 10 extensions) also includes access to VitXi.
Where can I report issues or suggest features for VitXi?
The VitalPBX community forum is the primary channel for bug reports, feature requests, and community discussion. The VitalPBX team actively monitors and participates in forum threads.
